Re: [OpenSIPStack] sip parser

[<jo...@op...>]

I am going to assume that you sub-classed the CallSession UACore for 
your gateway.  If this NOT what you did, then you need to give more 
info.  If this is what you did, all you need to do is to implement the 
two blank methods in CallSessionManager.  

BOOL CallSessionManager::OnRequireSDPAnswer(
  const SIPMessage & offer,
  SIPMessage & answer,
  CallSession & session
)
{
  /// Media UA should implement this method
  return FALSE;
}

BOOL CallSessionManager::OnRequireSDPOffer(
  SIPMessage & _offer,
  CallSession & session
)
{
  /// Media UA should implement this method
  return FALSE;
}


Lastly, as you can see, the default implementation of the CallSession 
does not have RTP. 

Joegen


haripriya alapati wrote:
>   
> Hi,
>
> I have already DSP and NETWORK modules. Only thing is I have to include the SIP module to my project.
> For Sip parser I am using opensipstack source code. But I am not getting how to separate the sip parser from the rtp and media modules in the opensipsatck code. 
> Can anyone please help me regarding this.
>
> Thanks in advance. 
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